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Network Working Group G. Camarillo
Request for Comments: 3486 Ericsson
Category: Standards Track February 2003
Compressing the Session Initiation Protocol (SIP)
Status of this Memo
This document specifies an Internet standards track protocol for the
Internet community, and requests discussion and suggestions for
improvements. Please refer to the current edition of the "Internet
Official Protocol Standards" (STD 1) for the standardization state
and status of this protocol. Distribution of this memo is unlimited.
Copyright Notice
Copyright (C) The Internet Society (2003). All Rights Reserved.
Abstract
This document describes a mechanism to signal that compression is
desired for one or more Session Initiation Protocol (SIP) messages.
It also states when it is appropriate to send compressed SIP messages
to a SIP entity.
Table of Contents
1. Introduction ............................................... 2
2. Overview of operation ...................................... 3
3. SigComp implementations for SIP ............................ 3
4. Sending a Request to a Server .............................. 3
4.1 Obtaining a SIP or SIPS URI with comp=sigcomp ........ 4
5. Sending a Response to a Client ............................. 5
6. Double Record-Routing ...................................... 6
7. Error Situations ........................................... 6
8. Augmented BNF .............................................. 7
9. Example .................................................... 7
10. Security Considerations .................................... 10
11. IANA Considerations ........................................ 10
12. Acknowledgements............................................ 10
13. Normative References ....................................... 10
14. Informative References ..................................... 11
15. Author's Address............................................ 11
16. Full Copyright Statement.................................... 12
1. Introduction
A SIP [1] client sending a request to a SIP server typically performs
a DNS lookup for the domain name of the server. When NAPTR [4] or
SRV [5] records are available for the server, the client can specify
the type of service it wants. The service in this context is the
transport protocol to be used by SIP (e.g., UDP, TCP or SCTP). A SIP
server that supports, for instance, three different transport
protocols, will have three different DNS entries.
Since it is foreseen that the number of transport protocols supported
by a particular application layer protocol is not going to grow
dramatically, having a DNS entry per transport seems like a scalable
enough solution.
However, sometimes it is necessary to include new layers between the
transport protocol and the application layer protocol. Examples of
these layers are transport layer security and compression. If DNS
was used to discover the availability of these layers for a
particular server, the number of DNS entries needed for that server
would grow dramatically.
A server that, for example, supported TCP and SCTP as transports, TLS
for transport security and SigComp for signaling compression, would
need the 8 DNS entries listed below:
1. TCP, no security, no compression
2. TCP, no security, SigComp
3. TCP, TLS, no compression
4. TCP, TLS, SigComp
5. SCTP, no security, no compression
6. SCTP, no security, SigComp
7. SCTP, TLS, no compression
8. SCTP, TLS, SigComp
It is clear that this way of using DNS is not scalable. Therefore,
an application layer mechanism to express support of signalling
compression is needed.
Note that for historical reasons both HTTP and SIP use a different
port for TLS on top of TCP than for TCP alone, although at
present, this solution is not considered scalable any longer.
A SIP element that supports compression will need to be prepared to
receive compressed and uncompressed messages on the same port. It
will perform demultiplexing based on the cookie in the topmost bits
of every compressed message.
2. Overview of operation
There are two types of SIP messages; SIP requests and SIP responses.
Clients send SIP requests to the host part of a URI and servers send
responses to the host in the sent-by parameter of the Via header
field.
We define two parameters, one for SIP URIs and the other for the Via
header field. The format of both parameters is the same, as shown in
the examples below:
sip:alice@atlanta.com;comp=sigcomp
Via: SIP/2.0/UDP server1.foo.com:5060;branch=z9hG4bK87a7;comp=sigcomp
The presence of this parameter (comp=sigcomp) in a URI indicates that
the request has to be compressed using SigComp, as defined in [2].
The presence of comp=sigcomp in a Via header field indicates that the
response has to be compressed using SigComp.
Therefore, the presence of comp=sigcomp indicates that the SIP entity
identified by the URI or by the Via header field supports SigComp and
is willing to receive compressed messages. Having comp=sigcomp mean
"willingness" as well as "support" allows the receiver of a SIP
message to influence the decision of whether or not to use SigComp at
a given time.
3. SigComp implementations for SIP
Every SIP implementation that supports SigComp MUST implement the
procedures described in this document.
4. Sending a Request to a Server
A request is sent to the host part of a URI. This URI, referred to
as the next-hop URI, is the Request-URI of the request or an entry in
the Route header field.
If the next-hop URI contains the parameter comp=sigcomp, the client
SHOULD compress the request using SigComp as defined in [2].
If the next-hop URI is a SIPS URI, the request SHOULD be compressed
before it is passed to the TLS layer.
A client MUST NOT send a compressed request to a server if it does
not know whether or not the server supports SigComp.
Regardless of whether the request is sent compressed or not, if a
client would like to receive subsequent requests within the same
dialog in the UAS->UAC direction compressed, this client SHOULD add
the parameter comp=sigcomp to the URI in the Contact header field if
it is a user agent client. If the client is a proxy, it SHOULD add
the parameter comp=sigcomp to its URI in the Record-Route header
field.
If a user agent client sends a compressed request, it SHOULD add the
parameter comp=sigcomp to the URI in the Contact header field. If a
proxy that Record-Routes sends a compressed request, it SHOULD add
comp=sigcomp to its URI in the Record-Route header field.
If a client sends a compressed request, it SHOULD add the parameter
comp=sigcomp to the topmost entry of the Via header field.
If a client does not know whether or not the server supports SigComp,
but in case the server supported it, it would like to receive
compressed responses, this client SHOULD add the parameter
comp=sigcomp to the topmost entry of the Via header field. The
request, however, as stated above, will not be compressed.
4.1 Obtaining a SIP or SIPS URI with comp=sigcomp
For requests within a dialog, a next-hop URI with the comp=sigcomp
parameter is obtained from a Record-Route header field when the
dialog is established. A client sending a request outside a dialog
can also obtain SIP URIs with comp=sigcomp in a Contact header field
in a 3xx or 485 response to the request.
However, clients establishing a session will not typically be willing
to wait until the dialog is established in order to begin compressing
messages. One of the biggest gains that SigComp can bring to SIP is
the ability to compress the initial INVITE of a dialog, when the user
is waiting for the session to be established. Therefore, clients
need a means to obtain a comp=sigcomp URI from their outbound proxy
before the user decides to establish a session.
One solution to this problem is manual configuration. However,
sometimes it is necessary to have clients configured in an automatic
fashion. Unfortunately, current mechanisms for SIP client
configuration (e.g., using DHCP [6]) do not allow to provide the
client with URI parameters. In this case, the client SHOULD send an
uncompressed OPTIONS request to its outbound proxy. The outbound
proxy can provide an alternative SIP URI with the comp=sigcomp
parameter in a Contact header field in a 200 OK response to the
OPTIONS. The client can use this URI for subsequent requests that
are sent through the same outbound proxy using compression.
RFC 3261 [1] does not define how a proxy should respond to an OPTIONS
request addressed to itself. It only describes how servers respond
to OPTIONS addressed to a particular user. Section 11.2 of RFC 3261
says:
Contact header fields MAY be present in a 200 (OK) response and
have the same semantics as in a 3xx response. That is, they may
list a set of alternative names and methods of reaching the user.
We extend this behavior to proxy servers responding to OPTIONS
addressed to them. They MAY list a set of alternative URIs to
contact the proxy.
Note that receiving incoming requests (even initial INVITEs)
compressed is not a problem, since user agents can REGISTER a SIP URI
with comp=sigcomp in their registrar. All incoming requests for the
user will be sent to this SIP URI using compression.
5. Sending a Response to a Client
A response is sent to the host in the sent-by parameter of the Via
header field. If the topmost Via header field contains the parameter
comp=sigcomp, the response SHOULD be compressed. Otherwise, the
response MUST NOT be compressed.
In order to avoid asymmetric compression (i.e., two SIP entities
exchanging compressed requests in one direction and uncompressed
requests in the other direction) proxies need to rewrite their
Record-Route entries in the responses. A proxy performing Record-
Route inspects the Record-Route header field in the response and the
Contact header field in the request that triggered this response (see
example in Section 9). It looks for the URI of the next upstream
(closer to the user agent client) hop in the route set. If this URI
contains the parameter comp=sigcomp, the proxy SHOULD add
comp=sigcomp to its entry in the Record-Route header field. If this
URI does not contain the parameter comp=sigcomp, the proxy SHOULD
remove comp=sigcomp (if it is present) from its entry in the Record-
Route header field.
The same way, a user agent server SHOULD add comp=sigcomp to the
Contact header field of the response if the URI of the next upstream
hop in the route set contained the parameter comp=sigcomp.
6. Double Record-Routing
Although proxies usually add zero or one Record-Route entries to a
particular request, some proxies add two of them to avoid Record-
Route rewriting. A typical example of double Record-Routing is a SIP
proxy that acts as a firewall between two networks. Depending on
which network a request comes from, it will be received on a
different interface by the proxy. The proxy adds one Record-Route
entry for one interface and a second one for the other interface.
This way, the proxy does not need to rewrite the Record-Route header
field on the response.
Proxies that receive compressed messages from one side of the dialog
(e.g., upstream) and uncompressed messages from the other side (e.g.,
downstream) MAY use the mechanism described above.
If a proxy detects that the next-hop proxy for a request is the proxy
itself and that the request will not be sent through the network, the
proxy MAY choose not to compress the request even if the URI contains
the comp=sigcomp parameter.
7. Error Situations
If a compressed SIP request arrives to a SIP server that does not
understand SigComp, the server will not have any means to indicate
the error to the client. The message will be impossible to parse,
and there will be no Via header field indicating an address to send
an error response.
If a SIP client sends a compressed request and the client transaction
times out without having received any response, the client SHOULD
retry the same request without using compression. If the compressed
request was sent over a TCP connection, the client SHOULD close that
connection and open a new one to send the uncompressed request.
Otherwise the server would not be able to detect the beginning of the
new message.
8. Augmented BNF
This section provides the augmented Backus-Naur Form (BNF) of both
parameters described above.
The compression URI parameter is a "uri-parameter", as defined by the
SIP ABNF (Section 25.1 of [1]):
compression-param = "comp=" ("sigcomp" / other-compression)
other-compression = token
The Via compression parameter is a "via-extension", as defined by the
SIP ABNF (Section 25.1 of [1]):
via-compression = "comp" EQUAL ("sigcomp" / other-compression)
other-compression = token
9. Example
The following example illustrates the use of the parameters defined
above. The call flow of Figure 1 shows an INVITE-200 OK-ACK
handshake between a UAC and a UAS through two proxies. Proxy P1 does
not Record-Route but proxy P2 does. Both proxies support
compression, but they do not use it by default.
UAC P1 P2 UAS
|(1)INVITE(c) | | |
|------------>| (2) INVITE | |
| |------------>| (3) INVITE |
| | |------------>|
| | | (4) 200 OK |
| | (5) 200 OK |<------------|
|(6)200 OK(c) |<------------| |
|<------------| | |
| | (7)ACK(c) | |
|-------------------------->| (8) ACK |
| | |------------>|
| | | |
| | | |
Figure 1: INVITE transaction through two proxies
Messages (1), (6) and (7) are compressed (c).
We provide a partial description of the messages involved in this
call flow below. Only some parts of each message are shown, namely
the Method name, the Request-URI and the Via, Route, Record-Route and
Contact header fields. We have not used a correct format for these
header fields. We have rather focus on the contents of the header
fields and on the presence (or absence) of the "comp=sigcomp"
parameter.
(1) INVITE UAS
Via: UAC;comp=sigcomp
Route: P1;comp=sigcomp
Contact: UAC;comp=sigcomp
P1 is the outbound proxy of the UAC, and it supports SigComp. The
UAC is configured to send compressed traffic to P1, and therefore, it
compresses the INVITE (1). In addition, the UAC wants to receive
future requests and responses for this dialog compressed. Therefore,
it adds the comp=Sigcomp parameter to the Via and to the Contact
header fields.
(2) INVITE UAS
Via: P1
Via: UAC;comp=sigcomp
Route: P2
Contact: UAC;comp=sigcomp
P1 forwards the INVITE (2) to P2. P1 does not use compression by
default, so it sends the INVITE uncompressed to P2.
(3) INVITE UAS
Via: P2
Via: P1
Via: UAC;comp=sigcomp
Record-Route: P2
Contact: UAC;comp=sigcomp
P2 forwards the INVITE (3) to the UAS. P2 supports compression, but
it does not use it by default. Therefore, it sends the INVITE
uncompressed. P2 wishes to remain in the signalling path and
therefore it Record-Routes.
(4) 200 OK
Via: P2
Via: P1
Via: UAC;comp=sigcomp
Record-Route: P2
Contact: UAS
The UAS generates a 200 OK response and sends it to the host in the
topmost Via, which is P2.
(5) 200 OK
Via: P1
Via: UAC;comp=sigcomp
Record-Route: P2;comp=sigcomp
Contact: UAS
P2 receives the 200 OK response. P2 Record-Routed, so it inspects
the Route set for this dialog. For requests from the UAS towards the
UAC (the opposite direction than the first INVITE), the next hop will
be the Contact header field of the INVITE, because P1 did not
Record-Route. That Contact identified the UAC:
Contact: UAC;comp=sigcomp
Since the UAC wants to receive compressed requests (Contact of the
INVITE), P2 assumes that the UAC would also like to send compressed
requests (Record-Route of the 200 OK). Therefore, P2 modifies its
entry in the Record-Route header field of the 200 OK (5). In the
INVITE (3), P2 did not used the comp=sigcomp parameter. Now it adds
it in the 200 OK (5). This will allow the UAC sending compressed
requests within this dialog.
(6) 200 OK
Via: UAC;comp=sigcomp
Record-Route: P2;comp=sigcomp
Contact: UAS
P1 sends the 200 OK (6) compressed to the UAC because the Via header
field contained the comp=sigcomp parameter.
(7) ACK UAS
Via: UAC;comp=sigcomp
Route: P2;comp=sigcomp
Contact: UAC;comp=sigcomp
The UAC sends the ACK (7) compressed directly to P2 (P1 did not
Record-Route).
(8) ACK UAS
Via: P2
Via: UAC;comp=sigcomp
Contact: UAC;comp=sigcomp
P2 sends the ACK (8) uncompressed to the UAS.
10. Security Considerations
A SIP entity receiving a compressed message has to decompress it and
to parse it. This requires slightly more processing power than only
parsing a message. This implies that a denial of service attack
using compressed messages would be slightly worse than an attack with
uncompressed messages.
An attacker inserting the parameter comp=sigcomp in a SIP message
could make a SIP entity send compressed messages to another SIP
entity that did not support SigComp. Appropriate integrity
mechanisms should be used to avoid this attack.
11. IANA Considerations
This document defines the "comp" uri-parameter and via-extension.
New values for "comp" are registered by the IANA at
http://www.iana.org/assignments/sip-parameters when new signalling
compression schemes are published in standards track RFCs. The IANA
Considerations section of the RFC MUST include the following
information, which appears in the IANA registry along with the RFC
number of the publication.
o Name of the compression scheme.
o Token value to be used. The token MAY be of any length, but
SHOULD be no more than ten characters long.
The only entry in the registry for the time being is:
Compression scheme Token Reference
--------------------- --------- ---------
Signaling Compression sigcomp RFC 3486
12. Acknowledgements
Allison Mankin, Jonathan Rosenberg and Miguel Angel Garcia-Martin
provided valuable comments on this memo.
13. Normative References
[1] Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,
Peterson, J., Sparks, R., Handley, M. and E. Schooler, "SIP:
Session Initiation Protocol", RFC 3261, June 2002.
[2] Price, R., Bormann, C., Christoffersson, J., Hannu, H., Liu, Z.
and J. Rosenberg, "Signaling Compression (SigComp)", RFC 3320,
January 2003.
[3] Bradner, S., "Key words for use in RFCs to indicate requirement
levels", BCP 14, RFC 2119, March 1997.
14. Informative References
[4] Mealling, M., "Dynamic Delegation Discovery System (DDDS) Part
Three: The Domain Name System (DNS) Database", RFC 3403, October
2002.
[5] Gulbrandsen, A., Vixie, P. and L. Esibov, "A DNS RR for
specifying the location of services (DNS SRV)", RFC 2782,
February 2000.
[6] Schulzrinne, H., "DHCP option for SIP servers", Work in
Progress.
15. Author's Address
Gonzalo Camarillo
Ericsson
Advanced Signalling Research Lab.
FIN-02420 Jorvas
Finland
EMail: Gonzalo.Camarillo@ericsson.com
16. Full Copyright Statement
Copyright (C) The Internet Society (2003). All Rights Reserved.
This document and translations of it may be copied and furnished to
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or assist in its implementation may be prepared, copied, published
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included on all such copies and derivative works. However, this
document itself may not be modified in any way, such as by removing
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Acknowledgement
Funding for the RFC Editor function is currently provided by the
Internet Society.

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